Displaying 20 results from an estimated 600 matches similar to: "asterisk-users Digest, Vol 70, Issue 25"
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 24
Yes this scenario works on my 2 systems which are at LAN. I made one system
as server (192.168.0.20) and registered from other system... it is fine but
now there is a different scene.
actually there is a registered user named abc at system1 (192.168.0.20)
having context [payasyougo] which is used to do outbound calls. we want to
use this user's context and account so that when we register
2010 May 11
1
asterisk-users Digest, Vol 70, Issue 23
Thanks Vardan,
I will like to know if this scenario can work when peer is not having fixed
ip and we use
host = nasir.server.com
?
also I have set insecure=invite,port
what if i use
insecure=no
thanks again.
Message: 24
Date: Tue, 11 May 2010 10:52:14 +0500
From: Vardan <hvardan71 at gmail.com>
Subject: Re: [asterisk-users] Dialing a SIP Peer without using
register strin
To:
2018 Oct 03
2
WebRTC as Softphone substitute ?
@Olivior
I agree that seting up WebRTC is hard, however when done it is smooth to
use. For replication you can build RPMs with working configurations.
Regarding stability, it is not being used widly, so can't say it is mature.
However we have no complain so far regarding audio or connectivity.
sometime we provide support for "allow media / mic" type issues, but you
know it is
2007 Jun 02
3
Dynamically adding Context in dialplan?
Hi everybody,
>From asterisk CLI we can add extensions in dial-plan dynamically using
"dialplan add extension" command.
but how we can dynamically create a context in dialplan. is that
possible?
Nasir Iqbal
2011 Mar 05
3
R Statistical Package Installation
Dear R-project team,
I have tried but could not install the R statistical package (http://cran.ms.unimelb.edu.au/ ) even after the help of my institute's IT personnel. The setup file could not be downloaded. The latest file R-2.12.2.tar.gz<http://cran.ms.unimelb.edu.au/src/base/R-2/R-2.12.2.tar.gz> does not start installation wizard. Kindly extend the technical support.
Best regards.
2010 Dec 20
5
DIALSTATUS on CANCEL
Hello,
We have a strange situation (asterisk 1.6.2.14), where we get a result for
DIALSTATUS for BUSY and No-ANSWER, but nothing for CANCEL.
This is the (relevant) test dialplan:
--------------------------------
[incoming-private]
exten => _X., n, Dial(SIP/1001,30)
exten => _X., n, NoOp(${DIALSTATUS})
exten => _X., n, Gosub(incoming-status,s-${DIALSTATUS},1)
[incoming-status]
exten
2018 Sep 29
2
WebRTC as Softphone substitute ?
Hi Olivior,
We have recently worked on a WebRTC based agent panel. As based on my
experience I think that WebRTC based phones are far better and cheaper then
those soft / sip phone. the big plus is that they are easy to customize and
developer can use the power of browser and web to build / offer features
which are not possible with regular phones.
Regarding your concern about BLF or call
2010 Jul 15
6
One way audio when dialing multiple registrations
Hi,
I am working on calling 2 registrations of same user on 2 different ip or
ports. It works fine and both phones ring simultaneously. the problem is
that there is one way audio, calling party can hear me but i can't hear
calling party.
here is the scenario..
SIP/XYZ at 192.168.0.20:5060
SIP/XYZ at 192.168.0.10:5678
i dial using following dial string
Dial(SIP/XYZ at
2010 May 18
3
About option U in Dial Ast version 1.6.2
Has any one used this?
U(x[^arg[^...]]):
x - Name of the subroutine to execute via Gosub
arg - Arguments for the Gosub routine
Execute via Gosub the routine <x> for the *called* channel before
connecting to the calling channel. Arguments can be specified to
the Gosub
using '^' as a delimiter. The Gosub routine can set the variable ${GO
2007 Jun 20
1
different codec for different extensions
Hi All,
I am wondering that how I can setup different codec for different
extensions in my dial plan.
scanario will
when user X (Sip) call 111 extension in default context. The Asterisk
response should be in GSM codec
When user X (Sip) call 222 extension in default context. the Asterisk
response should be in G711 Codec
Actually I want to setup an extension for FAX receiving (rx_fax) and
2010 Aug 03
2
RTP stream not passing through router with port forwarding
Hi,
I am trying to dial a registered user via his IP:Port mechanism, but problem
is that the audio data is not reaching to dialed user. here is the scenario.
caller and callee both are registered at asterisk server. asterisk server is
on public ip so no port forwarding and natting necessary there. however
caller and callee both are behind router and there is port forwarding
enabled and nat=yes,
2010 May 10
1
Dialing a SIP Peer without using register strin
Hi,
I am new to this list and this is first time i m posting here. please help
me out
currently I am working on dialing a sip peer on an asterisk server from 2nd
asterisk server. scenario is like this.
on my system i am using this peer in sip.conf.
[abc]
type=peer
username=abc
secret=mysecret
host=192.168.0.20
context=default
dtmfmode=rfc2833
;restrictcid=no
canreinvite=yes
2018 Oct 04
3
CURL to post application/json
We tried to use the CURL fn to POST json, but it's sent as form data and
there seems no support for changing the Content-Type header. We switched to
invoking curl in the shell.
All the documentation I could find says there is just one parameter for the
url and an optional second for POST body. Is there an undocumented way to
set Content-Type?
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2008 Oct 20
2
ISDN PRI Caller ID problem
Dear All,
I am trying to setup an ISDN line from local telco on a digium card. The
problem I am facing is that I am not getting any caller id from the
telco. They say that they have enabled caller id.
Please help me out.
My zapata.conf
--------------------------------------------------------------------------------------------------------------------
[trunkgroups]
[channels]
2010 May 13
0
asterisk-users Digest, Vol 70, Issue 30
sorry, you r right i just checked it with registration so there were astdb
entries for SIP registration.
anyhow after clearing settings frm astdb i tried the same scenario you
advised but no luck.
I think i told that i am not using server as peer but want to use a user
[abc] as peer so that when ever i use
dial(SIP/${EXTEN}@abc) or dial(SIP/abc/${EXTEN})
the call will be out from server using
2003 Sep 15
2
Unable to access the mailbox or folders !!
Hi all,
I have installed dovecot on redhat linux with
ldap backend. I can login using ldap account in to my
webmail (squirrelmail) .But when I login to the
webmail , I cant see any inbox or anything . Just some
error messages like this ,
ERROR:
ERROR : Connection dropped by imap-server.
Query: LIST "" "Sent"
ERROR : Could not complete request.
Query: SELECT
2010 May 21
3
CANCEL Reason
Hello all,
I need that Asterisk Always use Reason in a CANCEL.
How to do?
thank you
*Fran?ois *
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2007 May 30
12
False ring problem
Hi all,
when a user dials any number, asterisk automatically generates ringing which
caller can hear, and after 2 - 3 rings asterisk detects that the called user
is busy, then caller hears busy tone. for example user hears---
tone--tone--tobeep beep beep ---Can i some how eliminate the false ringing
at the start so that user hears only beep beep beep if the called user is
busy. I have used the R
2006 Apr 27
2
Asterisk Hangs the whole system
Is it possible for asterisk to hang the whole system ??
My Linux box is acting up, and I want to be sure which way to look.
Asterisk or some hardware.
--
Regards,
Nasir.
2007 May 31
5
Auto Dial Problem
Hi All,
I setup auto dial on my asterisk server. The problem
is asterisk does not wait for called party to answer
the call but proceed to process the extension specifed
in my .call file
My sample call file :
hannel: local/0124787924@outbound-reminder
MaxRetries: 5
RetryTime: 300
WaitTime: 40
Account: Reminder
context: remindem
extension: s
priority: 1
Set: MSG=0135.20070601.0124787924
Set: